2 * Copyright (c) 2013 Paul B Mahol
4 * This file is part of FFmpeg.
6 * FFmpeg is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Lesser General Public
8 * License as published by the Free Software Foundation; either
9 * version 2.1 of the License, or (at your option) any later version.
11 * FFmpeg is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Lesser General Public License for more details.
16 * You should have received a copy of the GNU Lesser General Public
17 * License along with FFmpeg; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21 #include "libavutil/avassert.h"
22 #include "libavutil/avstring.h"
23 #include "libavutil/opt.h"
24 #include "libavutil/samplefmt.h"
29 typedef struct AudioEchoContext {
31 float in_gain, out_gain;
32 char *delays, *decays;
37 int max_samples, fade_out;
41 void (*echo_samples)(struct AudioEchoContext *ctx, uint8_t **delayptrs,
42 uint8_t * const *src, uint8_t **dst,
43 int nb_samples, int channels);
46 #define OFFSET(x) offsetof(AudioEchoContext, x)
47 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
49 static const AVOption aecho_options[] = {
50 { "in_gain", "set signal input gain", OFFSET(in_gain), AV_OPT_TYPE_FLOAT, {.dbl=0.6}, 0, 1, A },
51 { "out_gain", "set signal output gain", OFFSET(out_gain), AV_OPT_TYPE_FLOAT, {.dbl=0.3}, 0, 1, A },
52 { "delays", "set list of signal delays", OFFSET(delays), AV_OPT_TYPE_STRING, {.str="1000"}, 0, 0, A },
53 { "decays", "set list of signal decays", OFFSET(decays), AV_OPT_TYPE_STRING, {.str="0.5"}, 0, 0, A },
57 AVFILTER_DEFINE_CLASS(aecho);
59 static void count_items(char *item_str, int *nb_items)
64 for (p = item_str; *p; p++) {
71 static void fill_items(char *item_str, int *nb_items, float *items)
73 char *p, *saveptr = NULL;
74 int i, new_nb_items = 0;
77 for (i = 0; i < *nb_items; i++) {
78 char *tstr = av_strtok(p, "|", &saveptr);
80 new_nb_items += sscanf(tstr, "%f", &items[i]) == 1;
83 *nb_items = new_nb_items;
86 static av_cold void uninit(AVFilterContext *ctx)
88 AudioEchoContext *s = ctx->priv;
92 av_freep(&s->samples);
95 av_freep(&s->delayptrs[0]);
96 av_freep(&s->delayptrs);
99 static av_cold int init(AVFilterContext *ctx)
101 AudioEchoContext *s = ctx->priv;
102 int nb_delays, nb_decays, i;
104 if (!s->delays || !s->decays) {
105 av_log(ctx, AV_LOG_ERROR, "Missing delays and/or decays.\n");
106 return AVERROR(EINVAL);
109 count_items(s->delays, &nb_delays);
110 count_items(s->decays, &nb_decays);
112 s->delay = av_realloc_f(s->delay, nb_delays, sizeof(*s->delay));
113 s->decay = av_realloc_f(s->decay, nb_decays, sizeof(*s->decay));
114 if (!s->delay || !s->decay)
115 return AVERROR(ENOMEM);
117 fill_items(s->delays, &nb_delays, s->delay);
118 fill_items(s->decays, &nb_decays, s->decay);
120 if (nb_delays != nb_decays) {
121 av_log(ctx, AV_LOG_ERROR, "Number of delays %d differs from number of decays %d.\n", nb_delays, nb_decays);
122 return AVERROR(EINVAL);
125 s->nb_echoes = nb_delays;
127 av_log(ctx, AV_LOG_ERROR, "At least one decay & delay must be set.\n");
128 return AVERROR(EINVAL);
131 s->samples = av_realloc_f(s->samples, nb_delays, sizeof(*s->samples));
133 return AVERROR(ENOMEM);
135 for (i = 0; i < nb_delays; i++) {
136 if (s->delay[i] <= 0 || s->delay[i] > 90000) {
137 av_log(ctx, AV_LOG_ERROR, "delay[%d]: %f is out of allowed range: (0, 90000]\n", i, s->delay[i]);
138 return AVERROR(EINVAL);
140 if (s->decay[i] <= 0 || s->decay[i] > 1) {
141 av_log(ctx, AV_LOG_ERROR, "decay[%d]: %f is out of allowed range: (0, 1]\n", i, s->decay[i]);
142 return AVERROR(EINVAL);
146 s->next_pts = AV_NOPTS_VALUE;
148 av_log(ctx, AV_LOG_DEBUG, "nb_echoes:%d\n", s->nb_echoes);
152 static int query_formats(AVFilterContext *ctx)
154 AVFilterChannelLayouts *layouts;
155 AVFilterFormats *formats;
156 static const enum AVSampleFormat sample_fmts[] = {
157 AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,
158 AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
163 layouts = ff_all_channel_layouts();
165 return AVERROR(ENOMEM);
166 ret = ff_set_common_channel_layouts(ctx, layouts);
170 formats = ff_make_format_list(sample_fmts);
172 return AVERROR(ENOMEM);
173 ret = ff_set_common_formats(ctx, formats);
177 formats = ff_all_samplerates();
179 return AVERROR(ENOMEM);
180 return ff_set_common_samplerates(ctx, formats);
183 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
185 #define ECHO(name, type, min, max) \
186 static void echo_samples_## name ##p(AudioEchoContext *ctx, \
187 uint8_t **delayptrs, \
188 uint8_t * const *src, uint8_t **dst, \
189 int nb_samples, int channels) \
191 const double out_gain = ctx->out_gain; \
192 const double in_gain = ctx->in_gain; \
193 const int nb_echoes = ctx->nb_echoes; \
194 const int max_samples = ctx->max_samples; \
195 int i, j, chan, av_uninit(index); \
197 av_assert1(channels > 0); /* would corrupt delay_index */ \
199 for (chan = 0; chan < channels; chan++) { \
200 const type *s = (type *)src[chan]; \
201 type *d = (type *)dst[chan]; \
202 type *dbuf = (type *)delayptrs[chan]; \
204 index = ctx->delay_index; \
205 for (i = 0; i < nb_samples; i++, s++, d++) { \
209 out = in * in_gain; \
210 for (j = 0; j < nb_echoes; j++) { \
211 int ix = index + max_samples - ctx->samples[j]; \
212 ix = MOD(ix, max_samples); \
213 out += dbuf[ix] * ctx->decay[j]; \
217 *d = av_clipd(out, min, max); \
220 index = MOD(index + 1, max_samples); \
223 ctx->delay_index = index; \
226 ECHO(dbl, double, -1.0, 1.0 )
227 ECHO(flt, float, -1.0, 1.0 )
228 ECHO(s16, int16_t, INT16_MIN, INT16_MAX)
229 ECHO(s32, int32_t, INT32_MIN, INT32_MAX)
231 static int config_output(AVFilterLink *outlink)
233 AVFilterContext *ctx = outlink->src;
234 AudioEchoContext *s = ctx->priv;
238 for (i = 0; i < s->nb_echoes; i++) {
239 s->samples[i] = s->delay[i] * outlink->sample_rate / 1000.0;
240 s->max_samples = FFMAX(s->max_samples, s->samples[i]);
241 volume += s->decay[i];
244 if (s->max_samples <= 0) {
245 av_log(ctx, AV_LOG_ERROR, "Nothing to echo - missing delay samples.\n");
246 return AVERROR(EINVAL);
248 s->fade_out = s->max_samples;
250 if (volume * s->in_gain * s->out_gain > 1.0)
251 av_log(ctx, AV_LOG_WARNING,
252 "out_gain %f can cause saturation of output\n", s->out_gain);
254 switch (outlink->format) {
255 case AV_SAMPLE_FMT_DBLP: s->echo_samples = echo_samples_dblp; break;
256 case AV_SAMPLE_FMT_FLTP: s->echo_samples = echo_samples_fltp; break;
257 case AV_SAMPLE_FMT_S16P: s->echo_samples = echo_samples_s16p; break;
258 case AV_SAMPLE_FMT_S32P: s->echo_samples = echo_samples_s32p; break;
263 av_freep(&s->delayptrs[0]);
264 av_freep(&s->delayptrs);
266 return av_samples_alloc_array_and_samples(&s->delayptrs, NULL,
272 static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
274 AVFilterContext *ctx = inlink->dst;
275 AudioEchoContext *s = ctx->priv;
278 if (av_frame_is_writable(frame)) {
281 out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
283 return AVERROR(ENOMEM);
284 av_frame_copy_props(out_frame, frame);
287 s->echo_samples(s, s->delayptrs, frame->extended_data, out_frame->extended_data,
288 frame->nb_samples, inlink->channels);
290 s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
292 if (frame != out_frame)
293 av_frame_free(&frame);
295 return ff_filter_frame(ctx->outputs[0], out_frame);
298 static int request_frame(AVFilterLink *outlink)
300 AVFilterContext *ctx = outlink->src;
301 AudioEchoContext *s = ctx->priv;
304 ret = ff_request_frame(ctx->inputs[0]);
306 if (ret == AVERROR_EOF && !ctx->is_disabled && s->fade_out) {
307 int nb_samples = FFMIN(s->fade_out, 2048);
310 frame = ff_get_audio_buffer(outlink, nb_samples);
312 return AVERROR(ENOMEM);
313 s->fade_out -= nb_samples;
315 av_samples_set_silence(frame->extended_data, 0,
320 s->echo_samples(s, s->delayptrs, frame->extended_data, frame->extended_data,
321 frame->nb_samples, outlink->channels);
323 frame->pts = s->next_pts;
324 if (s->next_pts != AV_NOPTS_VALUE)
325 s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
327 return ff_filter_frame(outlink, frame);
333 static const AVFilterPad aecho_inputs[] = {
336 .type = AVMEDIA_TYPE_AUDIO,
337 .filter_frame = filter_frame,
342 static const AVFilterPad aecho_outputs[] = {
345 .request_frame = request_frame,
346 .config_props = config_output,
347 .type = AVMEDIA_TYPE_AUDIO,
352 AVFilter ff_af_aecho = {
354 .description = NULL_IF_CONFIG_SMALL("Add echoing to the audio."),
355 .query_formats = query_formats,
356 .priv_size = sizeof(AudioEchoContext),
357 .priv_class = &aecho_class,
360 .inputs = aecho_inputs,
361 .outputs = aecho_outputs,